1. Field of the Invention
This invention relates to communications techniques, and more specifically, to systems and methods for increasing the effective data throughput of a transmission medium through the use of Hermite-Gaussian basis functions.
2. Description of Background Art
Years ago, the Internet was the domain of educators, scientists, military personnel, and technophiles. Web pages were utilitarian and simplistic. Many offered text-based information, or provided the most rudimentary of graphical interfaces. Expensive, elaborate equipment was required to access the Internet, but this hardware was purchased by large corporations or at governmental expense. By and large, the general consuming public did not have the means, inclination, or desire to access the Internet. But this has all changed.
More recently, the Internet has enjoyed an ever-expanding audience. In late 2002, it is something of a rarity to find a residential premises in the United States that does not have Internet access. A modem-equipped personal computer is almost as ubiquitous as a refrigerator, microwave oven, or a VCR (video cassette recorder). In the case of refrigerators, widespread usage does not pose an insurmountable technical hurdle, as the public utility company must merely increase its power generation capacity to keep up with the increased demand. However, in the case of the Internet, increased usage poses problems that are not so readily solved. Hundreds of thousands of individuals attempt to access the Internet every day. But they are not satisfied with text-based web pages or simplistic graphical interfaces—they would like to view real-time moving video images, listen to full-bandwidth audio, and download large files which may be several Megabytes in length.
Given the types of information that Internet users are presently accessing, heavy demands are placed on the bandwidth capacity of the user's Internet connection. Full-motion video, 20-20 kHz two-channel audio, large file downloads, and graphics-intensive websites require a bandwidth on the order of 200 Kilobytes to 3 Megabytes per second, give or take. On the other hand, most residential users access the Internet over a conventional subscriber loop to the local telephone company central office. In many cases, subscribers live more than two miles from the nearest central office, and cannot avail themselves of DSL service. Using state-of-the-art modem technology, this conventional (i.e., non-DSL) subscriber loop provides a bandwidth no greater than 56 Kilobytes per second and, in many cases, a lot less.
Expanded-bandwidth solutions exist on paper, but practical implementations of these solutions have not been realized. When a residential customer orders a service such as ISDN (Integrated Services Data Network) or DSL (Digital Subscriber Loop) from the local telephone company, it is difficult to obtain adequate performance. The services are theoretically available, but a residential customer requesting an ISDN line or DSL service often unwittingly becomes part of a “beta test”, due to lack of strong demand on the part of residential customers, and a consequent lack of practical experience on the part of the telephone company. Even if the various installation obstacles are eventually overcome, the monthly fees associated with an ISDN or DSL connection are cost-prohibitive for many customers.
Cable modems represent another alternative to conventional 56K modems for customers wishing to access the Internet. Nonetheless, telephone service is regarded as a basic necessity, present in virtually every household throughout the United States, whereas cable service is a non-essential or luxury item in locations with acceptable over-the-air reception. Moreover, since the FCC (Federal Communications Commission) has mandated conversion of analog over-the-air broadcasts to digital high-definition (HDTV) transmissions by 2006, many consumers will have access to acceptable over-the-air signals for the first time ever. With the recent proliferation of monthly subscription fees to cover everything from lawn maintenance to cellular telephone access, many customers would welcome the opportunity to avoid a monthly bill from the cable company.
An additional shortcoming of cable modems is that cable jacks are only provided in one or two rooms of a typical residential premises. If the customer desires Internet access from various locations throughout the home, this necessitates laborious cable-pulling through walls, floors, and attics. By contrast, telephone jacks are typically provided in a greater number of locations including bedrooms, kitchens, family rooms, basements, the garage, and even bathrooms.
What is needed is an improved method for accessing the Internet over a widely-available communications link. Such a method should not require the installation of additional wiring to a residential premises and, ideally, should operate over presently-existing communication paths. One possible candidate for such a communications path is the existing public switched telephone network (PSTN).
PSTN: The first PSTN communication system utilized twisted pairs of copper wire such that a single pair would carry one message at a time. Communications companies realized that, in order to enhance message-carrying capacity, they would need to devise techniques for transmitting several messages simultaneously over a single wire pair, because the cost of installing additional wires to accommodate increased demand was high. Companies that could reduce costs by putting more and more information over a single wire pair would have a competitive advantage.
Discoveries in transmission techniques enabled more than one message to be transmitted per wire pair, paving the way for the telegraph and telephone industry to become viable commercial enterprises. The challenge of maximizing effective bandwidth and increasing line capacity was present from the very beginning of telecommunications technology, and is still with us today.
Telecommunications networks provide the primary mechanism for conveying voice and data traffic between a source and a destination. But existing networks cannot handle the ever-increasing demand for capacity. Population increases, lower telephone rates, and increased data traffic over the Internet, all underscore the need to increase network capacity. But, as more and more bandwidth becomes available, higher bandwidth applications are quickly developed, such as high-resolution web pages and video-on-demand, which once again heightens the demand for increased bandwidth.
Any of several approaches could be employed to meet the increased demand for bandwidth. Additional transmission lines may be installed, additional satellites can be launched, and the radio spectrum can be more fully utilized. But all of these solutions are expensive and limited in scope. Satellites are launched into the Clarke Belt when it is desired to provide 24-hour service to a given geographical region. The Clarke Belt is a special location where satellites, when viewed from the Earth's surface, remain substantially stationary, permitting the use of conventional fix-mounted dish antennas. But, unfortunately, the Clarke Belt only has room for a limited number of satellites that can be placed in geostationary orbit. Wireless systems operate over the public radio spectrum, which, by its very nature, is a limited resource. Bandwidth utilization and compression methods may be employed to expand the capacity of wireless systems, but these methods are not sufficiently efficient to meet the demand in heavily-populated areas. To remain competitive, network service providers must endeavor to preserve the functionality of existing networks, yet still be able to accommodate the increasing bandwidth demand to handle voice, data, and video transmission.
In conventional analog transmission, voice energy acts to vibrate a diaphragm or crystal in a microphone, thereby converting mechanical vibrations into an electrical signal. The amplitude of this electrical signal varies in a manner analagous to the acoustical vibrations of the speaker's voice. This electrical signal can be amplified and transmitted over a wire pair to a receiver at a remote location. At the receiver, the electrical signal is used to energize an electromagnet, actuating a diaphragm in proximity to the magnet, whereby the diaphragm vibrates to reproduce the original voice. Digital transmission adds several steps to this transformation, for the voice is converted to an electrical current pattern whose varying amplitude is measured thousands of times per second. These measurements are encoded as voltage or amplitude levels, representing binary numbers consisting of “0” and “1”s.
Unlike analog transmission which conveys audio information as a continuous waveform, in digital transmission, binary numbers are transmitted in representational encoding schemes. Binary digits or bits may be transmitted singly, as discrete, on-off or zero/non-zero current pulses, or in groups as simultaneous pulses at different frequencies. At the receiving end, the bit stream is detected and used to modulate an analog current which drives a speaker. This method is “digital” because it entails conversion of an analog signal to numbers, and the transmission of digits in symbolic form.
Compression: Several existing methods provide for the transmission of information while reducing the overall bandwidth requirements. The most widely employed compression method uses mathematical algorithms and dictionary tables to manipulate and “point” digital signals in such a way that each transmission channel uses less bandwidth to carry recognizable information. Compression is achieved by building a predictive model of the waveform, removing unnecessary elements, and reconstructing the waveform from the remaining elements.
When converting an analog signal into digital form, accurate conversion requires at least twice as many measurements (samples per second), as the highest frequency in the signal. This sampling rate is oftentimes referred to as the Nyquist Criterion. The human voice generates sound frequencies in an approximate range of 20 to 4,000 Hz. Hence, a digital voice circuit, accepting an input in the range of 0-4,000 Hz, must sample this signal 8,000 times per second. In practice, the PSTN represents each sample using 8 bits of data. A single voice circuit, referred to as DS0, “digital signal level zero”, carries 64,000 (8,000×8) bits of data.
Compression methods are based upon reducing the number of bits required to convey a human voice or other data transmission. Currently-utilized compression algorithms can produce acceptable voice quality using less than 64 kbs by eliminating frequencies not necessary for voice intelligibility, particularly those below 300 Hz and those above 3,300 Hz, and possibly by emphasizing the frequencies in the 1,000 Hz range that carry most of the voice energy. Methods that drop an excessive amount of input signal tend to frustrate high-speed tonal data transmission schemes employed by modems and faxes. Currently-employed compression algorithms and equipment are able to transmit acceptable voice quality with a compression ratio of 8:1, using 8,000 bps per channel.
Using these compression methods, one channel can convey eight voice conversations over a line that originally was able to carry only one voice conversation. Higher compression methods which transmit voice over a circuit using less than 8,000 bps, suffer from increasing degradation of voice quality and “loss,” whereby at the receiving end of the line the voice in its original form is not clearly heard. Although new methods and algorithms may be employed to allow for clear voice transmission using less than 8,000 bps, there are appreciable limitations to these methods. All compression methods using algorithms suffer from greater and greater “loss” as compression ratios increase. Fax and video transmissions are more sensitive to bandwidth degradations and, hence, are more limited in their acceptable compression ratios.
While the main advantage of digital compression is increased network efficiency, in some situations, compression can degrade efficiency. For example, if the compression scheme is so complicated that it demands a significant amount of computer processing time to compress and decompress data, efficiency will suffer.
Multiplexing: One of the most widely-utilized forms of telecommunications service is known as the “T-1” protocol. T-1 uses a form of multiplexing in which 24 voice and/or data channels, each with 64,000 bps, can simultaneously exist on a pair of twisted copper wires. The total bandwidth capacity of T-1 is 1.544 Mbps. Compression methods are used in conjunction with T-1 and other transmission protocols to maximize bandwidth. Common compression systems using a ratio of 8:1, can carry 192 simultaneous voice or data channels (24×8) over a T-1 line.
Network service providers employ methods for increasing bandwidth through the utilization of compression and multiplexing, the most common multiplexing scheme in the United States being the T-1 protocol. Conversations or digital information carried on each T-1 line or channel is rendered unique, and transmitted with other channels over a common medium by multiplexing.
FDM (Frequency Division Multiplexing) has been used by phone companies to render each of a plurality of voice channels unique, when these channels are to be carried over a single transmission medium, such as a wire pair. Pursuant to one implementation of FDM, each of 24 voice and/or data channels are rendered distinct by having each channel assigned to a frequency band. For example, line 1 would use the frequency band of 0 Hz-4,000 Hz, line 2 would use the 4,000 Hz-8,000 Hz band, and so on. This method is best suited for analog signals which are subject to degradation and noise interference, and is not widely used at present. More common techniques are Time Division Multiplexing (TDM) and Statistical Multiplexing (STM), often called “Packet switching.”
Pursuant to TDM, each of the 24 channels (or lines) are rendered distinct by having each channel assigned to a particular, non-overlapping time slot. Frames of 24 time slots are transmitted, in which Channel 1 is allocated the first time slot in the frame, Channel 2 is allocated the second time slot, and so on. STDM works in a similar manner to TDM, assigning channels on the basis of time division. But STDM takes advantage of statistical fluctuations, and instead of automatically assigning each channel to a time slot, STDM assigns only active channels to time slots. Hence, instead of transmitting channels in sequential order (1, 2, 3, 4, 5, 6) as in TDM, STDM only assigns time slots to channels that are being used, e.g., 1, 3, 4, 5, 1, 6 etc. This method creates higher bandwidth utilization than TDM.